As the quality of network telephony systems has improved, there has been a migration of users from the traditional PSTN (Public Switched Telephone Network) to network telephony systems. With the proliferation of the Internet, Internet telephony has enabled distantly located users to communicate with one another using data protocols underlying the Internet. For example, the IP (Internet Protocol) suite along with various signaling protocols has made IP telephony a popular form of network telephony.
The Session Initiation Protocol (SIP) is a signaling protocol that may be used to assist with call set-up, management, and teardown. Other signaling protocols, such as the ITU-T H.323, MEGACO, and MGCP protocols, may also be used to implement various signaling functions. While these network telephony systems have provided advantages in cost and flexibility, certain challenges have arisen. In particular, network telephony systems are expected to enable users to easily move from one IP network to another.
One aspect of user mobility is to keep a call (or other session) connected while a user is moving from one domain to another. This first aspect is being addressed by efforts underlying the Mobile I.P. protocol, which is described in Perkins, C., “IP Mobility,” Internet Engineering Task Force (IETF) Request-For-Comments (RFC) 2002, October 1995, incorporated by reference herein.
Another aspect of user mobility is enabling a user to make and/or receive calls (or more generally to initiate and/or participate in a session) when the user moves from one domain to another without changing the user's identity. RFC 2002 does not adequately address this second aspect, nor does SIP or other signaling protocols.
Thus, a need exists for user mobility handling in a network telephony system.